Encoding and error correction system for enhanced performance of legacy communications networks

ABSTRACT

A system includes a header stripper configured to strip header data from a plurality of legacy system frames. Each of the plurality of legacy system frames (i) being in accordance with a legacy system frame format and (ii) including a header block and a traffic channel block. A first encoder is configured to encode speech data for a plurality of slots of the traffic channel blocks. A second encoder is configured to encode the stripped header data as a frame header. A combiner is configured to combine the frame header and the encoded speech data to generate a frame. A segmenter segments the frame into a plurality of segments. A transmitter is configured to transmit the plurality of segments as traffic channel data in accordance with the legacy system frame format.

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is a continuation of U.S. patent application Ser. No. 11/256,218, filed Oct. 21, 2005, now U.S. Pat. No. 7,712,005. The disclosure of the above application is incorporated herein by reference in its entirety.

FIELD OF THE INVENTION

This invention relates generally to the field of encoding and error correction for transmission systems and, more particularly, to use of an advanced vocoder with bit mapping and encoding for retrofit of legacy communications systems to enhance error correction performance.

BACKGROUND OF THE INVENTION

Waveform source coding and decoding (codec) is widely used in early digital mobile communication systems such as the Personal Handyphone System (PHS). Due to technology limitations at the time of implementation, some of the system designs did not provide appropriate channel encoding/decoding to protect the transferred data. For such systems, when the channel quality condition degrades, the ensuing high bit error rate makes voice performance unacceptable and some important control bits are easily corrupted. As the result, upper layer protocol and control mechanisms in the system would be likely active to turn off the channel. This is one of the most prevalent reasons for a lost connection during an ongoing communication session.

There are adaptive multi-rate (AMR) vocodec and corresponding channel coding capabilities in some advanced 2 G mobile systems and all the 3 G systems. Under the AMR standard, there are 8 different data rates for code Excited Linear Prediction (CELP) speech codec. These data rates range from 12.2 kbps to 4.75 kps. The more speech information is transferred, the better the voice performance is achieved. The basic approach employed in the AMR standard is that when the channel condition become worse, the system uses the modes with lower data transfer rate (of course, the voice performance is worse.). This saves more channel bandwidth and other resources for the system to increase the bit error correction ability. The lack comparable technology in legacy digital mobile systems (such as PHS) derives from under-developed algorithms and the expense of integrated circuit resources related to power and instruction speed requirements. With the advent of silicon technology, the use of digital signal processors (DSP) is no longer a luxury element in a PHS handset.

It is therefore desirable to make use of AMR vocodec capability in legacy systems.

It is also desirable to apply AMR and error correction in a manner which can be retrofit into early 2 G systems by re-arranging bit mapping to provide up to 6 to 7 dB gain for bit error reduction ability for certain AMR modes.

SUMMARY OF THE INVENTION

An encoding and error correction system and method according to the present invention employs the modern benefits of AMR codec by stripping header data from a plurality of legacy system frames having header and traffic channel (TCH) data blocks. Speech data is then encoded using the AMR to create bits for a data block substantially the same as contained in the plurality of frames. The stripped header data is encoded as a long frame header using a fixed convolution coder. The speech data is then convolutionally encoded and the long frame header and encoded speech data are combined as a long frame. The long frame is then deconstructed into a plurality of equal segments and the segments are transmitted as TCH data in the legacy system frame format.

BRIEF DESCRIPTION OF THE DRAWINGS

These and other features and advantages of the present invention will be better understood by reference to the following detailed description when considered in connection with the accompanying drawings wherein:

FIG. 1 is a block diagram of a prior art PHS system traffic channel frame and slot structure;

FIG. 2 is a block diagram of bit mapping and encoding for creation of a long frame for deconstruction into standard PHS slots for transmission;

FIG. 3 is a block diagram of the bit mapped long frame;

FIG. 4 is a block diagram of the Robust AMR Traffic Synchronized Control Channel (RATSCCH) format for the long frame;

FIG. 5 is a block diagram of an interleaving scheme for the encoded data;

FIG. 6 is a schematic diagram of the elements of a handset and base station system employing the present invention;

FIG. 7 a is a table representation of the AMR mode and division into Mode and class for use in the present invention; and,

FIG. 7 b is a flow chart representation of the mode switching algorithm using class as employed in the present invention.

DETAILED DESCRIPTION OF THE INVENTION

The present invention is defined for an exemplary embodiment employing the PHS communication system and standard (a 2 G legacy mobile system). The PHS system incorporating technology according to the invention will be referred to herein as Advanced PHS (APHS).

Data mapping of the Traffic Channel (TCH) of an exemplary PHS system is shown in FIG. 1. PHS is a time division multiplex (TDMA) system. One frame 10 is 5 ms in length and is divided into 8 slots, four for uplink and four for downlink. In each direction, three slots, T1, T2 and T3, are available for three different users and the last slot is the common control channel alternating uplink mode commands and downlink mode commands, C_up and C_down, for all three users.

Slot T2 is expanded in FIG. 1 as an example slot 12 and discussed in detail for the explanation of the invention herein. Areas PR and UW are employed for synchronization of the physical layer. Block UW incorporates 16 bits. As discussed in greater detail subsequently, PR and UW are inserted in the transmitted frame before the decoder so that they cannot be encoded. CI and SA are the protocol for slot format information and connection status and are important for connection reliability. Block CI contains 4 bits while block SA contains 16 bits. TCH contains the speech data and comprises 160 bits. The CRC block of 16 bits provides the error detection bits. It can be seen that within one frame, there are 160 bits of speech data and the whole vocodec rate is therefore 160/0.05=32 Kbps. This is the data rate for ADPCM (ITU G.726) as employed in PHS.

The present invention employs AMR codec in combination with bit mapping to create frames compatible with PHS transmission standards while adding encoding for performance enhancement. There are eight vocodec modes in an AMR system. The eight modes are defined by GSM and 3GPP international standards and the data rates employed are 12.2, 10.2, 7.95, 7.4, 6.7, 5.9, 5.15, and 4.75 (Kbps). Each of these rates is lower than the 32 Kbps basic capability of the PHS system and allows flexibility in formatting the data to be encoded.

As will be described in greater detail subsequently, the source speech and channel coding provided in the present invention is accomplished within 20 ms, or four PHS slot times. Interleaving of 20 ms blocks is beneath the sensitivity threshold of the human ear. The new encoding “frame” is referred to herein as a long-frame. To be compatible with the bit mapping of the PHS standard frame, the present invention does not make use of the CI, SA and CRC areas. That is, the encoded CI and SA data are put into the TCH block. The vocodec mode related message is encoded with a particular channel code mode and employs a special area and encoder. The larger the channel coding data block size, the better the result (higher bit error ability). Finally, a new control channel Robust AMR Traffic Synchronized Control Channel (RATSCCH) for long-frame synchronization is employed at the very beginning of the connection between of handset and the base station. RATSCCH is also used for a mode message in some special cases.

The TCH block is used for speech data, SA/CI and in-band data. Different vocodec modes have different speech and channel encoding parameters and require different data transfer rates. The SA/CI and in-band information are encoded and need a constant data rate. The encoding and bit mapping in APHS are shown in FIG. 2.

Four PHS slots 12 a, 12 b, 12 c and 12 d nominally provide the long frame. CI and SA data for the four slots are stripped and combined in an APHS long frame header block 14. Nominally 20 ms of speech samples 16 are processed through AMR vocodec 18. The long frame header data is processed through a first encoder 20 and in-band data 22 is inserted followed by the AMR processed speech samples which are routed through a second encoder 24 which is convolutional. The resulting long frame is shown in FIG. 3 wherein the header 26 comprises 156 bits of CI and SA data, an in-band block 28 carries 8 bits followed by an encoded speech data block 30 of 476 bits resulting in a long frame 32 of 640 bits. As previously described, this long frame is then split into four 160 bit lengths for insertion into the standard TCH block of 4 APHS frames 34 a, 34 b, 34 c, and 34 d for transmission.

In the PHS system, the data rate resource in area TCH for use in a long-frame is TCH*4=160*4=640 bits. For compatibility, the original area containing SA/CI data in each slot is reserved in APHS but the data is ignored for processing in the APHS. Message SA/CI together with in-band message data are encoded and put in original TCH area.

The maximum encoded speech data in a long-frame is 476 bits in the APHS embodiment described. Different vocode mode and channel code mode with different parameters are combined to generate different encoded speech data blocks of different size. If the generated encoded speech data block is larger than 476 bits, some bits have to be punctured. In APHS embodiment disclosed herein, channel coding is accomplished using a convolutional encoder. Other channel coding methods are employed in alternative embodiments.

Based on interleaving of the data, as will be discussed subsequently, the CI data need only be transmitted once in each long frame (the equivalent of one of every 4 PHS frames). The SA data needs to be transmitted for every slot. The resulting long frame data for SA/CI is shown in Table 1.

TABLE 1 Name SA0 SA1 SA2 SA3 Number of bits 16 16 16 16 Name CI Number of bits 4 Number of 6 CRC bits Polynomial D⁶ + D⁵ + D³ + D² + d¹ + 1 Number of 74 input bits for convolutional coder Polynomial for Polynomial ½ convolution G0/G0 = 1 coder G1/G0 = 1 + D + D³ + D⁴/1 + D³ + D⁴ with trailing 8 bits Number of 148 + 8 = 156 output bits

Eight bits are employed for mode information which is mapped into the final 8 bits of the long frame as will be shown below.

Table 2 shows the Codec mode and the associated convolution rate, the number of bits input into the convolutional coder, the resulting output number of bits from the coder, the number of SA bits after CRC and a convolutional encoder, the total number of bits and the preferred class.

TABLE 2 Number of output bits Number of Number from SA bits Total Number of input convolutional after CRC of bits for one bits to coder and 1/2 block 20 ms Codec convolutional (it should be convolution (632 + 8 = 640 Preferred mode Rate coder 476) coder total) Class TCH/AFS 12.2 1/2 250 508 156 632 1 puncturing 32 bits TCH/AFS 10.2 1/3 210 642 puncturing 156 632 2 166 bits TCH/AFS 7.95 1/3 165 513 puncturing 156 632 1 37 bits TCH/AFS 7.4 1/3 154 474 156 630 2 TCH/AFS 6.7 1/4 140 576 puncturing 156 632 1 100 TCH/AFS 5.9 1/4 124 520 puncturing 156 632 2 44 TCH/AFS 5.15 1/5 109 565 puncturing 156 632 1 89 TCH/AFS 4.75 1/5 101 535 puncturing 156 632 2 59

As can be seen in the table, with the fixed number of bits for SA/CI data of 156 and the total 640 bits available to fit within the PHS TCH of 160 bits for four frames, bits from the voice data convolutional encoder should total 476 and bits must be punctured to fit the long frame.

Exemplary convolution coding for each of the codec modes is shown in Tables 3—10 with definition of the punctured bits to maintain the long frame size of 640 bits.

TABLE 3 TCH/AFS12.2 Codec: The block of 250 bits {u(0)... u(249)} is encoded with the 1/2 rate convolutional code defined by the following polynomials: G0/G0 = 1 G1/G0 = 1 + D + D³ + D⁴/1 + D³ + D⁴ resulting in 508 coded bits, {C(0)... C(507)} defined by: r(k) = u(k) + r(k−3) + r(k−4) C(2k) = u(k) C(2k+1) = r(k)+r(k−1)+r(k−3)+r(k−4) for k = 0, 1, ..., 249; r(k) = 0 for k<0 and (for termination of the coder): r(k) = 0 C(2k) = r(k−3)+r(k−4) C(2k+1) = r(k)+r(k−1)+r(k−3)+r(k−4) for k = 250, 251, ..., 253 The code is punctured in such a way that the following 32 bits: C(417), C(421), C(425), C(427), C(429), C(433), C(437), C(441), C(443), C(445), C(449), C(453), C(457), C(459), C(461), C(465), C(469), C(473), C(475), C(477), C(481), C(485), C(489), C(491), C(493), C(495), C(497), C(499), C(501), C(503), C(505) and C(507) are not transmitted.

TABLE 4 TCH/AFS10.2 Codec: The block of 210 bits {u(0)... u(209)} is encoded with the 1/3 rate convolutional code defined by the following polynomials: G1/G3 = 1 + D + D³ + D⁴ / 1 + D + D² + D³ + D⁴ G2/G3 = 1 + D² + D⁴ / 1 + D + D² + D³ + D⁴ G3/G3 = 1 resulting in 642 coded bits, {C(0)... C(641)} defined by: r(k) = u(k) + r(k−1) + r(k−2) + r(k−3) + r(k−4) C(3k) = r(k) + r(k−1) + r(k−3) + r(k−4) C(3k+1) = r(k)+r(k−2)+r(k−4) C(3k+2) = u(k) for k = 0, 1, ..., 209 and (for termination of the coder): r(k) = 0 C(3k) = r(k)+r(k−1) + r(k−3) + r(k−4) C(3k+1) = r(k)+r(k−2)+r(k−4) C(3k+2) = r(k−1)+r(k−2)+r(k−3)+r(k−4) for k = 210, 211, ..., 213 The code is punctured in such a way that the following 22 bits: C(1), C(4), C(7), C(10), C(16), C(19), C(22), C(28), C(31), C(34), C(40), C(43), C(46), C(52), C(55), C(58), C(64), C(67), C(70), C(76), C(79), and C(82) are not transmitted. All these operations will result in 620 bits. The code is punctured in such a way that the following 166 bits: C(1), C(4), C(7), C(10), C(16), C(19), C(22), C(28), C(31), C(34), C(40), C(43), C(46), C(52), C(55), C(58), C(64), C(67), C(70), C(76), C(79), C(82), C(88), C(91), C(94), C(100), C(103), C(106), C(112), C(115), C(118), C(124), C(127), C(130), C(136), C(139), C(142), C(148), C(151), C(154), C(160), C(163), C(166), C(172), C(175), C(178), C(184), C(187), C(190), C(196), C(199), C(202), C(208), C(211), C(214), C(220), C(223), C(226), C(232), C(235), C(238), C(244), C(247), C(250), C(256), C(259), C(262), C(268), C(271), C(274), C(280), C(283), C(286), C(292), C(295), C(298), C(304), C(307), C(310), C(316), C(319), C(322), C(325), C(328), C(331), C(334), C(337), C(340), C(343), C(346), C(349), C(352), C(355), C(358), C(361), C(364), C(367), C(370), C(373), C(376), C(379), C(382), C(385), C(388), C(391), C(394), C(397), C(400), C(403), C(406), C(409), C(412), C(415), C(418), C(421), C(424), C(427), C(430), C(433), C(436), C(439), C(442), C(445), C(448), C(451), C(454), C(457), C(460), C(463), C(466), C(469), C(472), C(475), C(478), C(481), C(484), C(487), C(490), C(493), C(496), C(499), C(502), C(505), C(508), C(511), C(514), C(517), C(520), C(523), C(526), C(529), C(532), C(535), C(538), C(541), C(544), C(547), C(550), C(553), and C(556) are not transmitted.

TABLE 5 TCH/AFS7.95 Codec: The block of 165 bits {u(0)... u(164)} is encoded with the 1/3 rate convolutional code defined by the following polynomials: G4/G4 = 1 G5/G4 = 1 + D + D⁴ + D⁶ / 1 + D² + D³ + D⁵ + D⁶ G6/G4 = 1 + D + D² + D³ + D⁴ + D⁶ / 1 + D² + D³ + D⁵ + D⁶ generating 513 coded bits, {C(0)... C(512)} defined by: r(k) = u(k) + r(k−2) + r(k−3) + r(k−5) + r(k−6) C(3k) = u(k) C(3k+1) = r(k)+r(k−1)+r(k−4)+r(k−6) C(3k+2) = r(k)+r(k−1)+r(k−2)+r(k−3)+r(k−4)+r(k−6) for k = 0, 1, ..., 164; r(k) = 0 for k < 0 and (for termination of the coder): r(k) = 0 C(3k) = r(k−2) + r(k−3) + r(k−5) + r(k−6) C(3k+1) = r(k)+r(k−1)+r(k−4)+r(k−6) C(3k+2) = r(k)+r(k−1)+r(k−2)+r(k−3)+r(k−4)+r(k−6) for k = 165, 166, ..., 170 The code is punctured in such a way that the following 37 bits: C(1), C(2), C(4), C(5), C(8), C(22), C(70), C(118), C(166), C(214), C(262), C(310), C(317), C(319), C(325), C(332), C(334), C(341), C(343), C(349), C(356), C(358), C(365), C(367), C(373), C(380), C(382), C(385), C(389), C(391), C(397), C(404), C(406), C(409), C(413), C(415), and C(512) are not transmitted.

TABLE 6 TCH/AFS7.4 Codec: The block of 154 bits {u(0)... u(153)} is encoded with the 1/3 rate convolutional code defined by the following polynomials: G1/G3 = 1 + D + D³ + D⁴ / 1 + D + D² + D³ + D⁴ G2/G3 = 1 + D² + D⁴ / 1 + D + D² + D³ + D⁴ G3/G3 = 1 resulting in 474 coded bits, {C(0)... C(473)} defined by: r(k) = u(k) + r(k−1) + r(k−2) + r(k−3) + r(k−4) C(3k) = r(k) + r(k−1) + r(k−3) + r(k−4) C(3k+1) = r(k)+r(k−2)+r(k−4) C(3k+2) = u(k) for k = 0, 1, ..., 153 and (for termination of the coder): r(k) = 0 C(3k) = r(k)+r(k−1) + r(k−3) + r(k−4) C(3k+1) = r(k)+r(k−2)+r(k−4) C(3k+2) = r(k−1)+r(k−2)+r(k−3)+r(k−4) for k = 154, 155, ..., 157

TABLE 7 TCH/AFS6.7 Codec: The block of 140 bits {u(0)... u(139)} is encoded with the 1/4 rate convolutional code defined by the following polynomials: G1/G3 = 1 +D + D³ + D⁴ / 1 + D+D² + D³ + D⁴ G2/G3 = 1 + D² + D⁴ / 1 + D + D² + D³ + D⁴ G3/G3 = 1 G3/G3 = 1 producing 576 coded bits, {C(0)... C(575)} defined by: r(k) = u(k) + r(k−1) + r(k−2) + r(k−3) + r(k−4) C(4k) = r(k) + r(k−1) + r(k−3) + r(k−4) C(4k+1) = r(k)+r(k−2)+r(k−4) C(4k+2) = u(k) C(4k+3) = u(k)    for k = 0, 1, ..., 139; r(k) = 0 for k<0 Also (for termination of the coder): r(k) = 0 C(4k) = r(k)+r(k−1) + r(k−3) + r(k−4) C(4k+1) = r(k)+r(k−2)+r(k−4) C(4k+2) = r(k−1)+r(k−2)+r(k−3)+r(k−4) C(4k+3) = r(k−1)+r(k−2)+r(k−3)+r(k−4)    for k = 140, 141, ..., 143 The code is punctured in such a way that the following 100 bits: C(1), C(3), C(7), C(11), C(15), C(27), C(39), C(55), C(67), C(79), C(95), C(107), C(119), C(135), C(147), C(159), C(175), C(187), C(199), C(215), C(227), C(239), C(255), C(267), C(279), C(287), C(291), C(295), C(299), C(303), C(307), C(311), C(315), C(319), C(323), C(327), C(331), C(335), C(339), C(343), C(347), C(351), C(355), C(359), C(363), C(367), C(369), C(371), C(375), C(377), C(379), C(383), C(385), C(387), C(391), C(393), C(395), C(399), C(401), C(403), C(407), C(409), C(411), C(415), C(417), C(419), C(423), C(425), C(427), C(431), C(433), C(435), C(439), C(441), C(443), C(447), C(449), C(451), C(455), C(457), C(459), C(463), C(465), C(467), C(471), C(473), C(475), C(479), C(481), C(483), C(487), C(489), C(491), C(495), C(497), C(499), C(503), C(505), C(507), and C(511) are not transmitted.

TABLE 8 TCH/AFS5.9 Codec: The block of 124 bits {u(0)... u(123)} is encoded with the 1/4 rate convolution code defined by the following polynomials: G4/G6 = 1 + D² + D³ + D⁵ + D⁶ / 1 + D + D² + D³ + D⁴ + D⁶ G5/G6 = 1 + D + D⁴ + D⁶ / 1 + D + D² + D³ + D⁴ + D⁶ G6/G6 = 1 G6/G6 = 1 generating in 520 coded bits, {C(0)... C(519)} defined by: r(k) = u(k) + r(k−1) + r(k−2) + r(k−3) + r(k−4) + r(k−6) C(4k) = r(k) + r(k−2) + r(k−3) + r(k−5) + r(k−6) C(4k+1) = r(k) + r(k−1) + r(k−4) + r(k−6) C(4k+2) = u(k) C(4k+3) = u(k) for k = 0, 1, ..., 123; r(k) = 0 for k<0 and (for termination of the coder): r(k) = 0 C(4k) = r(k)+r(k−2) + r(k−3) + r(k−5) + r(k−6) C(4k+1) = r(k)+r(k−1)+r(k−4)+r(k−6) C(4k+2) = r(k−1)+r(k−2)+r(k−3)+r(k−4)+r(k−6) C(4k+3) = r(k−1)+r(k−2)+r(k−3)+r(k−4)+r(k−6) for k = 124, 125, ..., 129 The code is punctured in such a way that the following 44 bits: C(0), C(1), C(3), C(5), C(7), C(11), C(15), C(31), C(47), C(63), C(79), C(95), C(111), C(127), C(143), C(159), C(175), C(191), C(207), C(223), C(239), C(255), C(271), C(287), C(303), C(319), C(327), C(331), C(335), C(343), C(347), C(351), C(359), C(363), C(367), C(375), C(379), C(383), C(391), C(395), C(399), C(407), C(411), and C(415) are not transmitted.

TABLE 9 TCH/AFS5.15 Codec: The block of 109 bits {u(0)... u(108)} is encoded with the 1/5 rate convolution code defined by the following polynomials: G1/G3 = 1 + D + D³ + D⁴ / 1 + D + D² + D³ + D⁴ G1/G3 = 1 + D + D³ + D⁴ / 1 + D + D² + D³ + D⁴ G2/G3 = 1 + D² + D⁴ / 1 + D +D² + D³ + D⁴ G3/G3 = 1 G3/G3 = 1 generating 565 coded bits, {C(0)... C(564)} defined by: r(k) = u(k) + r(k−1) + r(k−2) + r(k−3) + r(k−4) C(5k) = r(k) + r(k−1) + r(k−3) + r(k−4) C(5k+1) = r(k) + r(k−1) + r(k−3) + r(k−4) C(5k+2) = r(k)+r(k−2)+r(k−4) C(5k+3) = u(k) C(5k+4) = u(k) for k = 0, 1, ..., 108; r(k) = 0 for k<0 and (for termination of the coder): r(k) = 0 C(5k) = r(k)+r(k−1) + r(k−3) + r(k−4) C(5k+1) = r(k)+r(k−1) + r(k−3) + r(k−4) C(5k+2) = r(k)+r(k−2)+r(k−4) C(5k+3) = r(k−1)+r(k−2)+r(k−3)+r(k−4) C(5k+4) = r(k−1)+r(k−2)+r(k−3)+r(k−4) for k = 109, 110, ..., 112 The code is punctured in such a way that the following 89 bits: C(0), C(4), C(5), C(9), C(10), C(14), C(15), C(20), C(25), C(30), C(35), C(40), C(50), C(60), C(70), C(80), C(90), C(100), C(110), C(120), C(130), C(140), C(150), C(160), C(170), C(180), C(190), C(200), C(210), C(220), C(230), C(240), C(250), C(260), C(270), C(280), C(290), C(300), C(310), C(315), C(320), C(325), C(330), C(334), C(335), C(340), C(344), C(345), C(350), C(354), C(355), C(360), C(364), C(365), C(370), C(374), C(375), C(380), C(384), C(385), C(390), C(394), C(395), C(400), C(404), C(405), C(410), C(414), C(415), C(420), C(424), C(425), C(430), C(434), C(435), C(440), C(444), C(445), C(450), C(454), C(455), C(460), C(464), C(465), C(470), C(474), C(475), C(480), and C(484) are not transmitted.

TABLE 10 TCH/AFS4.75 Codec: The block of 101 bits {u(0)... u(100)} is encoded with the 1/5 rate convolutional code defined by the following polynomials: G4/G6 = 1 + D² + D³ + D⁵ + D⁶ / 1 + D + D² + D³ + D⁴ + D⁶ G4/G6 = 1 + D² + D³ + D⁵ + D⁶ / 1 + D + D² + D³ + D⁴ + D⁶ G5/G6 = 1 +D + D⁴ + D⁶ / 1 + D + D² + D³ + D⁴ + D⁶ G6/G6 = 1 G6/G6 = 1 Generating 535 coded bits, {C(0)... C(534)} defined by: r(k) = u(k) + r(k−1) + r(k−2) + r(k−3) + r(k−4) + r(k−6) C(5k) = r(k) + r(k−2) + r(k−3) + r(k−5) + r(k−6) C(5k+1) = r(k) + r(k−2) + r(k−3) + r(k−5) + r(k−6) C(5k+2) = r(k) + r(k−1) + r(k−4) + r(k−6) C(5k+3) = u(k) C(5k+4) = u(k) for k = 0, 1, ..., 100; r(k) = 0 for k<0 and (for termination of the coder): r(k) = 0 C(5k) = r(k)+r(k−2) + r(k−3) + r(k−5) + r(k−6) C(5k+1) = r(k)+r(k−2) + r(k−3) + r(k−5) + r(k−6) C(5k+2) = r(k)+r(k−1)+r(k−4)+r(k−6) C(5k+3) = r(k−1)+r(k−2)+r(k−3)+r(k−4)+r(k−6) C(5k+4) = r(k−1)+r(k−2)+r(k−3)+r(k−4)+r(k−6) for k = 101, 102, ..., 106 The code is punctured in such a way that the following 59 bits: C(0), C(5), C(15), C(25), C(35), C(45), C(55), C(65), C(75), C(85), C(95), C(105), C(115), C(125), C(135), C(145), C(155), C(165), C(175), C(185), C(195), C(205), C(215), C(225), C(235), C(245), C(255), C(265), C(275), C(285), C(295), C(305), C(315), C(325), C(335), C(345), C(355), C(365), C(375), C(385), C(395), C(400), C(405), C(410), C(415), C(420), C(425), C(430), C(435), C(440), C(445), C(450), C(455), C(459), C(460), C(465), C(470), C(475), and C(479), are not transmitted.

The RATSCCH long frame employs a different format within the TCH block which is shown in FIG. 4. RATSCCH is used in two cases. At the very beginning of the connection between handset and base station, it is used for long-frame synchronization. Also, in some corner cases, the in-band message in RATSCCH is used to give the encoder mode information together with the in-band message of the normal frame. RATSCCH as used in the present invention is comparable to the formats widely used in GSM/3G systems for telling remote PS/CS to change AMR mode and class.

In one embodiment, interleaving of the data of the long frame for transmission in the standard PHS 5 ms bursts is accomplished as shown in FIG. 5. The bits of the long frame are interleaved as shown in Table 11.

TABLE 11 Interleaving Table for PHS Load K = 0 1 2 3 4 5 6 7 0 513 442 307 172 37 550 415 0 64 577 506 371 236 101 614 479 1 128 57 570 435 300 165 30 543 2 192 121 634 499 364 229 94 607 3 256 185 50 563 428 293 158 23 4 320 249 114 627 492 357 222 87 5 384 313 178 43 556 421 286 151 6 448 377 242 107 620 485 350 215 7 512 441 306 171 36 549 414 279 8 576 505 370 235 100 613 478 343 9 56 569 434 299 164 29 542 407 10 120 633 498 363 228 93 606 471 11 184 49 562 427 292 157 22 535 12 248 113 626 491 356 221 86 599 13 312 177 42 555 420 285 150 15 14 376 241 106 619 484 349 214 79 15 440 305 170 35 548 413 278 143 16 504 369 234 99 612 477 342 207 17 568 433 298 163 28 541 406 271 18 632 497 362 227 92 605 470 335 19 48 561 426 291 156 21 534 399 20 112 625 490 355 220 85 598 463 21 176 41 554 419 284 149 14 527 22 240 105 618 483 348 213 78 591 23 304 169 34 547 412 277 142 7 24 368 233 98 611 476 341 206 71 25 432 297 162 27 540 405 270 135 26 496 361 226 91 604 469 334 199 27 560 425 290 155 20 533 398 263 28 624 489 354 219 84 597 462 327 29 40 553 418 283 148 13 526 391 30 104 617 482 347 212 77 590 455 31 168 33 546 411 276 141 6 519 32 232 97 610 475 340 205 70 583 33 296 161 26 539 404 269 134 63 34 360 225 90 603 468 333 198 127 35 424 289 154 19 532 397 262 191 36 488 353 218 83 596 461 326 255 37 552 417 282 147 12 525 390 319 38 616 481 346 211 76 589 454 383 39 32 545 410 275 140 5 518 447 40 96 609 474 339 204 69 582 511 41 160 25 538 403 268 133 62 575 42 224 89 602 467 332 197 126 639 43 288 153 18 531 396 261 190 55 44 352 217 82 595 460 325 254 119 45 416 281 146 11 524 389 318 183 46 480 345 210 75 588 453 382 247 47 544 409 274 139 4 517 446 311 48 608 473 338 203 68 581 510 375 49 24 537 402 267 132 61 574 439 50 88 601 466 331 196 125 638 503 51 152 17 530 395 260 189 54 567 52 216 81 594 459 324 253 118 631 53 280 145 10 523 388 317 182 47 54 344 209 74 587 452 381 246 111 55 408 273 138 3 516 445 310 175 56 472 337 202 67 580 509 374 239 57 536 401 266 131 60 573 438 303 58 600 465 330 195 124 637 502 367 59 16 529 394 259 188 53 566 431 60 80 593 458 323 252 117 630 495 61 144 9 522 387 316 181 46 559 62 208 73 586 451 380 245 110 623 63 272 137 2 515 444 309 174 39 64 336 201 66 579 508 373 238 103 65 400 265 130 59 572 437 302 167 66 464 329 194 123 636 501 366 231 67 528 393 258 187 52 565 430 295 68 592 457 322 251 116 629 494 359 69 8 521 386 315 180 45 558 423 70 72 585 450 379 244 109 622 487 71 136 1 514 443 308 173 38 551 72 200 65 578 507 372 237 102 615 73 264 129 58 571 436 301 166 31 74 328 193 122 635 500 365 230 95 75 392 257 186 51 564 429 294 159 76 456 321 250 115 628 493 358 223 77 520 385 314 179 44 557 422 287 78 584 449 378 243 108 621 486 351 79

For interleaving, the bits are split into even and odd bits 102, 104 and interleaved according to the table. Eight segments 106 of 80 bits of data are obtained 108 which are then interleaved with eight segments from the prior 20 ms segment 110. After interleaving, two segments at a time are transmitted in the TCH block of the PHS slot.

In the normal data transfer mode, the system operates in a similar manner to AMR in GSM and 3GPP systems. As shown in FIG. 6. In the handset 36, speech encoder 38 for input voice samples the data and provides coded data to a channel encoder 40 which converts the long frames to PHS format 5 ms frames for transmission. Uplink speech data is then transmitted to the base station 42 in which a channel decoder 44 collapses the PHS standard frames into the long frame format which is then passed to the speech decoder 46. During the decoding, the bit error and corresponding SNR (relative channel condition) is estimated by the Viterbi decoder in the convolution decoding process. After estimation within a certain time period, the codec adaption unit 48 makes decision whether the encode mode should be changed in the receiving direction, as will be described in detail subsequently. A request message, uplink node command 50, is inserted into the in-band area and transfer to the handset.

Operation of downlinking data from the base station to the handset is comparable with the speech data encoded by speech encoder 52 for input voice samples the data and provides coded data to a channel encoder 54 which converts the long frames to PHS format 5 ms frames for transmission. Downlink speech data is then transmitted to the handset in which a channel decoder 56 collapses the PHS standard frames into the long frame format which is then passed to the speech decoder 58. During the decoding, the bit error and corresponding SNR (relative channel condition) is estimated. After estimation within a certain time period, the codec adaption unit 60 makes decision whether the encode mode should be changed in the receiving direction. A request message, downlink mode command 62, is inserted into the in-band area and transfer to the base station.

For each transmitted long-frame, the encode mode (index) of the transmitter is determined based on the mode command which shifts the mode using an algorithm based on class as shown in FIGS. 7 a and 7 b. PHS standard requirements allow a simplified algorithm for mode adjustment over standard GSM/3G systems. Adjacent modes are in different classes effectively splitting the total number of modes as shown in the table of FIG. 7 a. As shown in FIG. 7 b, if the transmission is proceeding in mode 11 and class 1 (12.2 kbps) and the error estimate from the convolution decoder indicates a reduction in bit rate is required, the mode command causes a shift 72 of one mode down within the same class, i.e. to mode 10 class 1 (7.95 kbps). If the BER is still too high, a second shift 74 down is commanded to mode 01 class 1 (6.7 kbps). Continuing high BER results in yet another shift 76 down to mode 00 class 1 (5.15 kbps). If the BER remains too high, a final shift 78 across class to mode 00 class 2 (4.74 kbps) is made. Thus the entire range of available modes is spanned in four command shifts. Similarly, if BER is reduced below a predefined threshold, the command shifts up the class for increased transmission speed. For example if the system has been in lowest mode, i.e. mode 00 class 2 (4.74 kbps) and BER is now improving, a command shift 80 raises the mode to mode 01 class 2 (5.9 kbps) as opposed to returning to mode 00 class 1. Further improvement results in a shift 82 to mode 10 class 2 (7.4 kbps), with yet further improvement resulting in a shift 84 to mode 11 class 2 (10.2 kbps). If further improvement is available, the algorithm then causes the mode to shift class 86, i.e. to mode 11 class 1 (12.2 kbps) for highest transmission speed. Improvement or deterioration in BER in the middle of a class range results in movement up or down within the class as exemplified by shifts 88 and 90. The algorithm allows a circular shifting pattern based on mode within the class, shifting class only when reaching a minimum or maximum transmission capability within the class.

Having now described the invention in detail as required by the patent statutes, those skilled in the art will recognize modifications and substitutions to the specific embodiments disclosed herein. Such modifications are within the scope and intent of the present invention as defined in the following claims. 

What is claimed is:
 1. A system comprising: a header stripper configured to strip header data from a plurality of legacy system frames, each of the plurality of legacy system frames (i) being in accordance with a legacy system frame format and (ii) including a header block and a traffic channel block; a first encoder configured to encode speech data for a plurality of slots of the traffic channel blocks; a second encoder configured to encode the stripped header data as a frame header; a combiner configure to combine the frame header and the encoded speech data to generate a frame; a segmenter to segment the frame into a plurality of segments; and a transmitter configured to transmit the plurality of segments as traffic channel data in accordance with the legacy system frame format.
 2. The system of claim 1, wherein the first encoder comprises an adaptive multi-rate codec, wherein the adaptive multi-rate codec is configured to encode the speech data.
 3. The system of claim 2, wherein the adaptive multi-rate codec is configured to select a codec mode based on channel quality.
 4. The system of claim 3, wherein the adaptive multi-rate codec selects the codec mode by: at least one of (i) shifting down from a first mode to a second mode within a first class and (ii) shifting down from the first mode to a third mode in a second class upon degradation of a bit error rate; and at least one of (i) shifting up from the first mode to a fourth mode within the first class and (ii) shifting up from the first mode to a fifth mode of the second class upon improvement of the bit error rate.
 5. The system of claim 1, wherein the stripped header data comprises channel identification and slow associated control data.
 6. The system of claim 1, wherein the second encoder comprises a convolution encoder configured to encode the stripped header data.
 7. The system of claim 6, wherein the convolution encoder comprises a ½ convolution encoder.
 8. A system comprising: a header stripper configured to strip headers from a first plurality of slots of data, wherein each of the first plurality of slots of data comprise a header block and a traffic channel block; a first encoder configured to combine and encode the headers of the first plurality of slots of data to generate a frame header; a second encoder configured to combine and encode the traffic channel blocks of the first plurality of slots of data to generate data samples; a third encoder configured to combine and encode the frame header and the data samples to generate an encoded frame; a segmenter configured to segment the encoded frame into a second plurality of slots of data; and a transmitter configured to transmit the second plurality of slots of data.
 9. The system of claim 8, wherein the first plurality of slots of data and the second plurality of slots of data are personal handy phone system slots of data.
 10. The system of claim 8, wherein the first plurality of slots of data and the second plurality of slots of data are in a legacy system frame format.
 11. The system of claim 8, wherein the combiner is configured to: combine the frame header, in-band data, and the data samples to generate a frame; and encode the frame to generate the encoded frame.
 12. The system of claim 8, wherein the second encoder includes an adaptive multi-rate encoder.
 13. The system of claim 8, wherein the third encoder includes a convolutional encoder. 